I highly recommend "Asterisk Hacking" as well.
http://www.amazon.com/Asterisk-Hacking-Ben-Jackson/dp/1597491519
Tim Nelson
Systems/Network Support
Rockbochs Inc.
- Original Message -
From: "Bill Andersen" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing L
the major manufacturers not support this? If
you're using groundstart, what hardware are you using? Thank you!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
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asterisk-users
ifferentiate itself from the ULTRA MEGA HUGE telco
QUEST.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
- Original Message -
From: "Ignacio Ortega A." <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, Mar
My first thought looking at the site was "SCAM"!!! maybe my second thought
would be "SCRAM" ... is this company even "legit"
On Fri, Mar 21, 2008 at 10:35 PM, Tim Nelson < [EMAIL PROTECTED] > wrote:
Apparently the list description of "Non
I may be missing something here... but won't a 32bit binary run just fine on a
64bit platform? Would you even see a performance increase or advantage to a
64bit soft phone versus a 32bit version?
Tim Nelson
Systems/Network Support
Rockbochs Inc.
- Original Message -
From: "zo
http://svn.digium.com/view/zaptel?view=rev&revision=4121
Pure VoIP is the wave of the future!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
Disclaimer: We all know what day it is today... :-)
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ycom IP430 handsets almost
exclusively for this installation. Can anyone think of a reason why a call
would randomly go on hold?
Tim Nelson
Systems/Network Support
Rockbochs Inc.
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ve voltage that is too
low...
Tim Nelson
Systems/Network Support
Rockbochs Inc.
- Original Message -
From: "Todd" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, April 1, 2008 8:20:36 PM (GMT-0600) America/Chicago
. When anyone answers the call, all they hear is silence. The
calling party that called in on ZAP/5 still hears the line ringing. It is very
odd.
We're using Asterisk 1.2.12.1, Zaptel 1.2.12, and Wanpipe 3.2.1 on CentOS 4.4.
Thank you for your help!!!
Tim Nelson
Systems
d has fantastic support.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
- Original Message -
From: "Danny Lockard" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, May 23, 2008 11:54:09 AM GMT -06:00 US/Canada
It looks like you may be missing a context declaration right after your
"channel => 1" line. Try adding "context=incoming" right after that.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Original Message -
From: "Drew Gibson"
You are correct... my mistake. :-/
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Original Message -
From: "Drew Gibson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, June 6, 2008 1:36:
diagnose
and repair the problem? Thank you!!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
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To UNSUBSCRIBE or update options v
Well... the problem magically cleared itself up this morning. Nothing was
changed in the meantime. I'd love to know what the problem was...
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Original Message -
From: "Steve Totaro" <[EMAIL PROTECT
ifference.
All suggestions welcome!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Original Message -----
From: "Tim Nelson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, June 13, 2008 2:54:2
will
need a molex power connector plugged in to provide extra voltage. With
traditional phone systems, I have not seen any distance limitations. I would
just like to verify that the FXS ports are able to provide sufficient power for
longer runs. Thank you!!!
Tim Nelson
Systems/Network Su
otice that unless you had
all the phones in a single room for testing... :-) In production with one phone
in each room in scattered locations, it should not be an issue.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332
- Original Message -
From: "Forrest Beck" &l
of disk activity would
cause little blips and beeps in the audio stream. Make sure you have all
extraneous unneeded devices turned off in the BIOS.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332
- Original Message -
From: "Steve Totaro" <[EMAIL PROTECTED]&g
Could you possibly post what steps you took to make this work so others
(including myself :-) ) may benefit? Thank you!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332
- Original Message -
From: "Michael Munger" <[EMAIL PROTECTED]>
To: "Asterisk Use
It appears that the number you're dialing (701) is not an extension or UA but
rather a call parking slot.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332
- Original Message -
From: "Sanjoy Rath" <[EMAIL PROTECTED]>
To: "Asterisk Users Mai
Even if * is shutdown, zaptel is still running and your ISDN channels are still
technically up. Shutting down zaptel should close the channels and put those
circuits into alarm mode.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332
- Original Message -
From
ation that is
understandable for the task. Any help is greatly appreciated. Thank you!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
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asterisk-users mailing list
To UNSUBSCRI
sed/placed/received calls. I would
think that simply remapping one of those keys should not pose a problem.
Thank you for the Polycom admin guide reference. I'll continue my search there.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
- Original Message -
From: "Jose Quinteir
nk? We are running Asterisk 1.2.12.1 and
Zaptel 1.2.22.1. Any ideas?!?
Tim Nelson
Systems/Network Support
Rockbochs Inc.
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Thank you all for the suggestions. I'm looking into getting groundstart lines
for that installation as suggested earlier. Also, I'll try setting the outbound
call routes in reverse from the inbound hunt group. I appreciate your help!
Tim Nelson
Systems/Network Support
Roc
:-) HAHA.. Unfortunately, PRI service is not available at this location...
Thank you for the help!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
- Original Message -
From: "Jay R. Ashworth" <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Sent: Thursday, Feb
Yes... this installation has a Sangoma A400D card fully populated. Thanks again.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
- Original Message -
From: "John Novack" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
S
- Original Message -
> http://camrivox.com/products/flexor-cti-salesforce/
>
> We've used this for a few clients.
>
How were your experiences with it? I have a customer that will want this type
of integration in the near future, and would love to hear how installation,
operation, and s
Greetings-
I recently experienced an odd situation. I have an Asterisk 11.5.0 system (Box
A) with a SIP peering to another Asterisk 1.8.23.0 system (Box B). At some
point, Box A started sending over 65Mbps of SIP OPTIONS packets to Box A. I do
have qualify=yes for the peer on both sides, and th
- Original Message -
> SIP options message is due to check the peer registration is
> keepalive. As per my understanding it might be because of network
> flap may be wireshark trace can give you any clue.
> Regards
Correct. I understand the role and function of the OPTIONS requests. The
- Original Message -
> On 13 Feb 2014, at 18:10, Tim Nelson wrote:
> > I recently experienced an odd situation. I have an Asterisk 11.5.0
> > system (Box A) with a SIP peering to another Asterisk 1.8.23.0
> > system (Box B). At some point, Box A started sending o
Greetings-
As many of your are Polycom "experienced", I was hoping some kind soul could
provide direction on a specific issue.
On a system running Asterisk 11.11.0 (and latest FreePBX), I'm finding an
instance where, using intercom/paging functionality of FreePBX, I need to
override an end u
- Original Message -
> Tim,
> I THINK but I'm not sure that you can do this with the Polycom
> multicast page function. Have you attempted this yet?
> Thanks
> david
Given the odd nature of multicast paging with Polycom, I was hoping to avoid
such a setup. My recollection is having thi
Greetings-
Working with the T.38 gateway functionality that is sparsely documented [1] ,
I'm attempting to get the following functional:
Asterisk calling system -> Asterisk system in T.38 Gateway Mode (box in
question) -> SIP Provider
The problem is:
-The provider is not initiating a rein
- Original Message -
> Greetings-
> Working with the T.38 gateway functionality that is sparsely
> documented [1], I'm attempting to get the following functional:
> Asterisk calling system -> Asterisk system in T.38 Gateway Mode (box
> in question) -> SIP Provider
> The problem is:
>
- Original Message -
> On 10/22/2014 03:55 PM, Tim Nelson wrote:
> > - Original Message -
> >
> >> Greetings-
> >
> >> Working with the T.38 gateway functionality that is sparsely
> >> documented [1], I'm attempting to get the f
- Original Message -
>
>
> On 23/10/2014 3:55 AM, Tim Nelson wrote:
> > - Original Message -
> >
> >> Greetings-
> >
> >> Working with the T.38 gateway functionality that is sparsely
> >> documented [1], I'm attempting
- Original Message -
>
>
> On 23/10/2014 10:07 PM, Larry Moore wrote:
> >
> >
> > On 22/10/2014 11:23 AM, Tim Nelson wrote:
> >> Greetings-
> >>
> >> Working with the T.38 gateway functionality that is sparsely
> >> docu
- Original Message -
>
>
> On 22/10/2014 11:23 AM, Tim Nelson wrote:
> > Greetings-
> >
> > Working with the T.38 gateway functionality that is sparsely
> > documented
> > [1], I'm attempting to get the following functional:
> >
- Original Message -
> Hi
>
> Thank you for your support.
> The server is actually compromised, I discovered that after making a
> deep trace using the audit daemon and looking for the kill signal
> (SIGKILL) that terminates asterisk.
> I discovered that there is an executable with a rand
Take a look at 'screen'. Chances are, it's already installed on your boxen.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Original Message -
From: "Joe Pukepail" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List
I've also seen systems where the IRQ between the card and another heavily
loaded device (disk controller) are shared causing clicks, beeps, and pops to
be present in the audio stream.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- "Alan Lord" &l
Yep... your A104d has HWEC onboard (as signified by the 'd' on your model). It
is necessary to set echocancel=no and probably echocancelwhenbridged=no in your
zapata.conf to get reliable faxing to work.
Tim Nelson
Systems/Network Engineer
Rockbochs Inc.
(218)727-4332 x105
- &
OpenVPN?
--Tim
- "Erick Perez" wrote:
> Excuse my ignorance but if i have an asterisk in a LAN, and i have
> users in their homes/internet (dozens), in order to correctly connect
> those users across my firewall, what is the technology that i need to
> buy, called?
> secure border gateway?
You guys think YOU'RE overdoing it... your solution works with a single line.
My solution was some convoluted 100 line shell script!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- "Lenz Emilitri" wrote:
>
I have a feeling we'
you have any sort of site/mailing list/etc setup to facilitate this group?
I'd be interested in attending such a meetup in the future.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- "asterisk help" wrote:
> Hello Asterisk Users and those with an Int
Is your soft phone also using rfc2833 for DTMF mode?
--Tim
- "David @ULC" wrote:
>
> IVR Number :17275691533
> When I try it from xlite configuring my provider directly, it works
> perfectly.
>
> When I try to dial out from dialer , it doesnt work.
> [sip8]
> type=peer
> username=
ing a full blown asterisk install
may be overly complicated for what you need?
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
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asterisk-users mailing list
Another war dialer with IAX capabilities:
http://www.softwink.com/iwar/
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- "Steve Edwards" wrote:
> This may be of interest -- as a tool we can use to test our systems
> and as
> a weapon that may
- "Jon Pounder" wrote:
> Tim Nelson wrote:
>
Now you're making others think I wrote this tirade of crap.
> The fact that this would be even being discussed on this list is an
> embarrassment to the asterisk community.
>
Isn't this the *perfect*
Did your iax.conf get overwritten with the upgrade?
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- "Carlos Chavez" wrote:
> I just upgraded a very old Asterisk installation to the last 1.2.31 I
> can find in Asterisk.org site. Now for some reaso
- ameu...@yahoo.fr wrote:
>
> I have to develop a VoIP application. I need to know how to use Java APIs to
> communicate to my client application with asterisk.
I tried looking for some answers based upon your subject but nothing came up.
This may be what you're looking for: http://lmgtfy
- "Wilton Helm" wrote:
>
>If half-duplex audio is good enough for you, sure.
You've lost me there. I am not aware of a modem that is for sale today that is
half duplex. (OK some support a couple of minor half duplex modes). All state
of the art modem protocols send and receive simultaneo
e
> phone company.
>
> Perhaps munge the ring tone detect if nothing else?
>
> Cary
>
Greetings Cary-
I had the same situation a while back. Please see my post and the answer from
another kind user here:
http://lists.digium.com/pipermail/asterisk-users/2009-January/224545.h
A good place to start is here:
http://www.venturevoip.com/news.php?rssid=1513
FreePBX includes a module called 'Zoip' which allows you to play Zork via a
Text-to-speech engine.
Why on Earth someone would want to do so is beyond me but hey... why not. :-)
Tim Nelson
Syste
While I'm not sure this is the source of your problems, I've seen it ruin
otherwise acceptable SIP situations.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
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- "James Mutuku" wrote:
> Hellos,
>
I want to setup Asterisk+a2billing for over 10,000 extensions for voip resale.
Has anyone done this before. What are the hardware requirements and challenges?
>
James
If you're asking that sort of question, you probably shouldn't be doing it. The
Greetings!
I'm hoping someone can help me with what should be the most basic of problems.
Essentially, I want to have certain calls on an Asterisk 1.2.25 (Yes I know its
old, upgrade, etc... its on my roadmap) install go out a couple of analog lines
and all other calls go out a PRI. The analog
ap/g0.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim
> Nelson
> Sent: Tuesday, May 19, 2009 10:38 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subje
hen through the PRI ? Is that why you're putting the priority 101 in
> the PRI context ?
>
> Martin
>
> On Tue, May 19, 2009 at 10:37 AM, Tim Nelson
> wrote:
> > Greetings!
> >
> > I'm hoping someone can help me with what should be the most basic o
p/g0.
> >
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim
> Nelson
> > Sent: Tuesday, May 19, 2009 10:38 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discuss
- "Jonathan Moore" wrote:
> I've got an Asterisk server, and several SIP phones behind our router
> here. Things are working just perfectly inside the network, just as
> the should.
>
> However, I'm not trying to configure my asterisk server to talk with
> SIP services outside our network, o
"Farooq Hussain" wrote:
>
Hello Everyone,
I am receiving following error message will making Zaptel on Cent OS 5.2.
make[1]: Entering directory `/usr/src/zaptel-1.4.12.1'
> echo "You do not appear to have the sources for the 2.6.18-92.el5 kernel
> installed."
> You do not appear to h
you want but you'll need to do some
heavy sifting as it contains very verbose output.
Also, post this on the Trixbox forums. I'm sure someone over there can lend a
hand also.
Good luck!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- "Douglas Mo
[EMAIL PROTECTED] ~]# asterisk -rx 'sip show channels'
assuming you want SIP... substitute sip for iax2 if you prefer...
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- "nik600" <[EMAIL PROTECTED]> wrote:
> Hi to all.
>
> Is poss
'oh323 show channels' I would assume... I don't have a box handy with h323
loaded to verify.
Check http://astrecipes.net/index.php?n=89
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- "nik600" <[EMAIL PROTECTED]> wrote:
> A
here aren't any system resource constraints
such as high CPU usage or memory usage... :-)
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- "c james" <[EMAIL PROTECTED]> wrote:
> A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
&g
What is the call volume on this box? Depending on the version of Asterisk,
maybe there are some memory leaks present causing calls to fail but everything
else to keep working?
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- "Gregory Malsack" wrote:
Tell your PRI provider that you want one of those channels exclusively bound to
your fax DID. Also, it should be removed from the normal hunt group where the
rest of your calls come in. Then, the only way that PRI channel will be used is
when someone calls your fax number/DID.
Tim Nelson
I've worked with many providers who are able to do this. In fact, we're using
such a setup on our office PRI. I'm not sure how they're achieving this on
their end however...
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- "Andrew Thomas"
g these in a number of
schools that have Rauland paging systems and they work wonderfully.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- "Gordon Henderson" wrote:
> On Mon, 22 Dec 2008, Jerry Geis wrote:
>
> > Is there an ATA type device out th
Check out this howto: http://engineertim.com/?p=16
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- "Michiel van Baak" wrote:
> On 11:04, Fri 09 Jan 09, Matthew Nicholson wrote:
> > On Fri, 2009-01-09 at 16:49 +, Steve Howes wrote:
> >
Greetings list-
I have a box with a single FXO card in it. I'm able to dial out ZAP/1 with no
problems and as expected. However, I would like inbound calls on that POTS line
to go unanswered by Asterisk since I have other equipment on the line. I've
setup zapata.conf for the channel without a c
> -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-
> > boun...@lists.digium.com] On Behalf Of Tim Nelson
> > Sent: Sunday, January 11, 2009 12:35 AM
> > To: asterisk-users@lists.digium.com
> > Subject: [asterisk-users] Use ZA
oun...@lists.digium.com
> [mailto:asterisk-users-
> > boun...@lists.digium.com] On Behalf Of Tim Nelson
> > Sent: Sunday, January 11, 2009 12:35 AM
> > To: asterisk-users@lists.digium.com
> > Subject: [asterisk-users] Use ZAP/Dahdi channel for outbound
> only...
> >
27;re suffering from clipping of some sort or almost a
half-duplex audio situation. Weird. I've seen some el-cheapo cordless phones
behave this way but never a 'business' solution. Eek.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- "Darrick H
rtainly was not
coming into the conversation as an expert, just stating what I'd read/heard of
their service... hence the "My understanding is that..." beginning to the
email. :-)
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
_
promised and hosed with a rootkit. If you find odd processes, weird files,
getting segfaults for normal applications (bash/grep/vim/etc) then you'll
likely want to backup your data and reinstall from scratch. Please post here if
you have further problems/questions.
Good luck!
Tim Ne
Greetings list-
I'd like the ability to hangup all calls for a particular extension from the
system CLI. I understand this can probably be scripted using the AMI but I'm
not familiar on how to do it. Help!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727
I used the AMI show the current channels, grep out the ones I wanted to kill,
and then used Command: Hangup to get rid of them. And, all done with about 100
lines of nasty nasty bash. :-)
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- "Alexander Lopez&qu
- Original Message -
> I use the latest spandsp source from the freeswitch git.
> There you have also a changelog documenting the differences. Steve
> Underwood
> commit here the latest changes in spandsp source.
>
> http://fisheye.freeswitch.org/changelog/freeswitch.git/libs/spandsp
>
D
Greetings-
I currently have a customer that *requires* key-system functionality in an
Asterisk PBX. On a SIP phone, the BLF keys need to show the current state of
the analog lines attached to the system (DAHDI FXO). By pressing one of these
keys (for line 1 for example), the dialed number needs
- Original Message -
> On 01/26/2012 09:46 AM, Tim Nelson wrote:
> > Greetings-
> >
> > I currently have a customer that *requires* key-system functionality
> > in an Asterisk PBX. On a SIP phone, the BLF keys need to show the
> > current state of the an
- Original Message -
>
> Yes, this is exactly what I am looking for - hopefully in English :-)
>
>
> Date or range selection would make this perfect. I have been looking
> for something like this for quite a while but there is none. I would
> really appreciate it if you share this with m
- Original Message -
> Hello Folks;
>
> I know this is a non-commercial discussion group, but I am looking for
> some open-source software suggestions
>
>
> We are going to be setting up a prepaid PBX service with the following
> features:
>
>
> • Email to Fax and Fax to Email
> • Inwa
- Original Message -
> Hi,
>
> When someone says "T.38 is not reliable on a (normally loaded and
> managed) LAN", would you rather agree or disagree ?
> In this case, fax calls are coming in through an analog gateway,
> passing trough Asterisk and then going out to ISDN through a digital
>
Greetings-
First off, my apologies for the slightly OT nature of this post. It does
involve Asterisk to a degree, but errs a bit on the side of Audiocodes inquiry.
I accept all responsibility for my actions and the consequences. :)
The scenario is this: I have an Asterisk box connected to a Med
- Original Message -
> Greetings-
>
> First off, my apologies for the slightly OT nature of this post. It
> does involve Asterisk to a degree, but errs a bit on the side of
> Audiocodes inquiry. I accept all responsibility for my actions and the
> consequences. :)
>
> The scenario is this
Greetings-
I've had reports of a customer PBX acting strangely to some inbound calls.
Specifically, a call comes into an FXO port, hits a Dial() to ring a few
extensions, but by the time someone answers the phone, the call has been
dropped, and the caller is listening to on-hold music. There is
Bart Coninckx wrote:
> Hi all,
>
> for smaller (or maybe even bigger) sites I'm looking for a smaller,
> appliance-type like PC, preferably solid state and fanless PC.
> Since it's only going to run Asterisk for a couple of extensions I
> don't think CPU and RAM need to be maxed out.
>
> Does anyon
- Original Message -
> On Thursday 10 May 2012, Bart Coninckx wrote:
> > I'm looking for a smaller,
> > appliance-type like PC, preferably solid state and fanless PC.
> > Since it's only going to run Asterisk for a couple of extensions I
> > don't
> > think CPU and RAM need to be maxed out.
- Original Message -
> Tim,
>
> looked at these briefly, they all seemed pre-installed, correct? Is
> reinstallation with, let's say, CentOS possible?
>
> thx,
>
> BC
>
The units *can* come preinstalled with our PBX flavor (Debian, Asterisk,
FreePBX), or they can be sent bare and you
- Original Message -
> On 05/10/2012 03:49 AM, Bart Coninckx wrote:
> > Hi all,
> >
> > for smaller (or maybe even bigger) sites I'm looking for a smaller,
> > appliance-type like PC, preferably solid state and fanless PC.
> > Since it's only going to run Asterisk for a couple of extensions
- Original Message -
> On 05/10/12 18:38, Kevin P. Fleming wrote:
> > On 05/10/2012 03:49 AM, Bart Coninckx wrote:
> >> Hi all,
> >>
> >> for smaller (or maybe even bigger) sites I'm looking for a smaller,
> >> appliance-type like PC, preferably solid state and fanless PC.
> >> Since it's o
- Original Message -
> Hi,
>
> I'm facing a strange situation.
> Though it's not directly related to Asterisk, I do think it is
> interesting to this mailing list.
>
>
> The setup is a single line which is split between an ADSL
> modem/routeur and a fax machine (Asterisk was removed from
- Original Message -
> Hi Steve,
>
> you are telling me there is no way to set a particular speed on my
> iaxmodem in order to force the sender speed?
> I have some problems with a customer who gets malformed faxes even if
> no
> error occurs. Since I cannot tell the sender to lower its fa
- Original Message -
> - Original Message -
> > Hi Steve,
> >
> > you are telling me there is no way to set a particular speed on my
> > iaxmodem in order to force the sender speed?
> > I have some problems with a customer who gets malformed faxes even
> > if
> > no
> > error occur
- Original Message -
> Hi guys, thanks for answers.
>
> That could seem counter-intuitive but it is not. Not to mention the
> fact
> that information technology is not science,
Huh? It is indeed very much a science. You have known established facts,
processes, concepts, methods for test
- Original Message -
> On 05/17/2012 07:53 AM, Andrew Furey wrote:
> > we use ActiveFax for sending (interfaced from an ERP package) and
> > often get Comm Error 283 and incomplete faxes. If it's just making
> > a
> > bad situation worse, how is it that our solution of turning off ECM
> > m
- Original Message -
> On 05/23/2012 08:41 PM, Cody Harris wrote:
> > Hello All,
> > I use IAX2 as the incoming connection from my DID provider. For
> > whatever reason, this works best for me, SIP connections lag very
> > frequently and only have about a 50% success rate for incoming
> >
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